Recently, there has been in practical use a communication system which sends to a high-bit-rate digital line a signal obtained by compressing and multiplexing speech signals. Conventionally, speech signals are multiplexed at a bit rate of 64 Kbps, and a multiplexed signal thus obtained is sent to a high-bit-rate digital line. Further, recently there has also been provided a system in which speech signals have been multiplexed at a bit rate of 32 Kbps.
FIG. 1 shows a conventional communication system having a function of compressing and multiplexing speech signals in order to send these signals to a high-bit-rate digital line. The communication system shown in FIG. 1 functions as a WAN (Wide Area Network), and comprises exchanges 10A, 10B and 10C, such as PBXs (Private Branch Exchanges), transmission devices 20A, 20B and 20C, and transmission paths L1 and L2 which are high-bit-rate digital lines. The transmission path L1 connects the transmission devices 20A and 20B to each other, and the transmission line L2 connects the transmission devices 20B and 20C to each other.
Each of the exchanges 10A, 10B and 10C accommodates terminals T1, T2 and T3, such as telephone sets, and terminals other than telephone sets, such as data terminals and facsimile machines A circuit of a communication path system of each of the exchanges 10A, 10B and 10C switches 64 Kbps signals. Hence, each of the exchanges 10A, 10B and 10C has a PCM (Pulse Code Modlation) converter which converts analog signals from the terminals T1, T2 and T3 into a PCM speech signal having a bit rate of 64 Kbps and executes the reverse operation. It may be possible to provide the above PCM converter for each analog terminal.
The transmission device 20A connected to the exchange 10A comprises a compressor 21, an expander 22 and a multiplexing circuit 23. The compressor 21 converts the 64 Kbps PCM speech signal into an ADPCM (Adaptive Differential Pulse Code Modulation) signal having a bit rate of 32 Kbps. As shown, the compressor 21 is comprised of an adaptive quantizer 21a, a predictor 21b and a difference device 21c. Generally, a plurality of compressors 21 are provided. The multiplexing circuit 23 multiplexes the 32 Kbps ADPCM signals with other digital signals (image signals and data signals), and sends a multiplexed signal to the transmission path L1. Further, the multiplexing circuit 23 separates a plurality of digital signals from a multiplexed signal received from the transmission line L1. The expander 22 demodulates the received 32 Kbps ADPCM signal to generate a 64 Kbps PCM speech signal. The demodulated PCM speech signal is sent to the exchange 10A. The compressor 21 and the expander 22 form a speech codec. The transmission device 20C and the exchange 10C connected thereto are respectively configured in the same manner as the above transmission device 20A and the exchange 10A.
The transmission device 20B comprises a multiplexing circuit 24 connected to the transmission line L1, and a speech codec 25 connected thereto. Further, the transmission device 20B comprises a multiplexing circuit 27 connected to the transmission line L2, and a speech codec 26 connected thereto. Each of the speech codecs 25 and 26 has the functions of both the compressor 21 and the expander 22.
It will now be assumed that the terminal T1 accommodated in the exchange 10A generates a call requesting to communicate with the terminal accommodated in the exchange 10B, a speech signal from the terminal T1 being converted into a 64 Kbps PCM speech signal in the exchange 10A, which is then converted into a 32 Kbps ADPCM speech signal by the compressor 21 of the transmission device 20A. The ADPCM signal is multiplexed with other signals, and then sent to the transmission path L1. The multiplexing device of the transmission device 20B demultiplexes the received multiplexed signal to generate separate signals. The codec 25 converts the 32 Kbps ADPCM signal into the 64 Kbps PCM speech signal. The exchange 10B discriminates a dial signal in a control channel received via the multiplexing device 24, and recognizes that the call from the terminal T1 should terminate at the terminal T2. The exchange 10B specifies a channel coupling the terminals T1 and T2 to each other, and then they become connected to each other.
In a case where the terminal T1 calls the terminal T3 accommodated in the exchange 10C, the exchange 10B analyzes a dial signal from the terminal T1 and recognizes that the call should be sent to the exchange 10C. In this case, the exchange 10C sets a channel connecting the transmission paths L1 and L2 to each other. A signal received from the transmission path L1 via the multiplexing circuit 24 and the speech codec 25 is sent to the transmission path L2 via the exchange 10B, the speech codec 26 and the multiplexing circuit 27. During this time, the 64 Kbps PCM speech signal from the exchange 10B is converted into the 32 Kbps ADPCM signal by the speech codec 26. Then, the speech signal from the terminal T1 is supplied to the terminal via the transmission device 20C and the exchange 10C. A speech signal from the terminal T3 is transmitted to the terminal T1 in a route the reverse of the above route
As described above, in the communication system, the compression and expansion processes are repeatedly carried out. The above bit-rate conversion (64 Kbps.fwdarw.32 Kbps) does not cause a deterioration of speech in terms of the principle. The compression and expansion process using the ADPCM is particularly called transcoding. That is, on the transmission side, the next sampling value is predicted from a sampling value, and only the difference between the predicted value and the real value is quantized and transmitted. On the receiving side, a predicted value is added to the transmitted residual signal to reproduce the real value. On the transmission side, the quantizing step is changed so that the difference between the predicted value and the real value becomes smaller. By using the above transcoding technique, the quality of the 32 Kbps speech signal on the transmission path L1 is theoretically the same as that of the 32 Kbps speech signal on the transmission path L2 even when the above compression and expansion processes are repeatedly carried out. As a result, there is no deterioration of speech even when the speech signal is repeatedly relayed and transmitted.
However, a problem will occur when the transmission paths L1 and L2 transmit further compressed signals, such as 16 Kbps or 8 Kbps signals. Generally, in a speech signal up to a low bit rate lower than or equal to 16 Kbps, it is very difficult to apply the transcoding technique used in the 32 Kbps ADPCM. That is, the quality of speech will greatly deteriorate if the low-bit-rate signals having a bit rate lower than or equal to 16 Kbps are processed by the transcoding technique.
In a conventional technique, a process shown in FIG. 2 is used in order to transmit the speech signal at a low bit rate lower than or equal to 16 Kbps. In the transmission device 20B, surplus bits (dummy bits) amounting to 48 Kbps (56 Kbps) are added to the 16 Kbps (8 Kbps) speech signal received from the transmission path L1, so that a 64 Kbps signal is generated a shown in FIG. 2). The 64 Kbps signal containing the dummy bits is switched via a time switch TS without being changed, and sent to the transmission device 20B b in FIG. 2). The transmission device 20B removes the dummy bits from the received 64 Kbps signal and reproduces the 16 Kbps (8 Kbps) speech signal. When the 16 Kbps (8 Kbps) signal should be only switched for relay, it is sent to an objective path (transmission path L2 in this case) c in FIG. 2). On the other hand, when the 16 Kbps (8 Kbps) signal is a call requested to terminate at the terminal T2, the 16 Kbps (8 Kbps) speech signal is output to the speech codec having the 16 Kbps.fwdarw.64 Kbps conversion function in the transmission device 20B (c in FIG. 2). The codec converts only the 16 Kbps speech signal into a 64 Kbps speech signal, and outputs the 64 Kbps speech signal to a line switch LS in the exchange 10B (d in FIG. 2). The line switch LS outputs the 64 Kbps speech signal to the terminal T2.
However, the structure shown in FIG. 2 has a disadvantage in that the efficiency of processing is poor and it takes a long time to process the signals because irrespective of whether or not each call should be relayed to the transmission path L2 or terminate at a terminal accommodated in the exchange 10B, the dummy bits are added to all input speech signals and removed there from and thereafter the 16 Kbps (8 Kbps) speech signals are converted into the 64 Kbps signals which can be processed by the exchange 10B.